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Routing between Legacy PBX and Multiple Trixbox

aulax
Posts: 9
 Member Since:
2007-02-08
Submitted by aulax on Fri, 02/09/2007 - 9:10am.

Here is what we have configured:

3 - Avaya Definity G3si PBX currently with IP trunking and 4 digit dialing between regional offices, extensions 5xxx, 41xx, 34xx

3 - Trixbox PBX's currently setup at 3 other office locations, extensions 36xx, 37xx, and 38xx. All are interconnected via SIP trunks to allow for 4 digit dialing between those offices.

1 - Trixbox as bridge between Avaya and other deployed Trixbox's. Connected via TE110P using all 23 b-channels, 1 d-channel. This is also connected with the other Trixbox's with SIP trunks. I can setup an extension and call via 4 digits to any of the other Trixbox's.

Network configuration is hub - spoke model, with regional offices connected via private point-to-point lines to corporate HQ.

Currently, from my Avaya desk phone, I can call 5xxx, 41xx, and 34xx without a problem; they route to respective G3si and complete the call. From any of the Trixbox's I can call any extension on either the Avaya or Trixbox and they complete successfully. However, when I try to dial a Trixbox extension from an Avaya extension, the call routes to the bridge and dies. From the Avaya side, I can see the call route to the T1 tie line, and be passed to the Trixbox Bridge successfully. I can see the call in the Call Log on the Trixbox, it has the destination of the extension I am calling, but it will not route to the appropriate Trixbox and complete the call.

What am I missing?

‹ Problems downloading ISO's recorded calls in mp3 format ›

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dickson
Posts: 1831
 Member Since:
2006-06-02
Re: Routing between Legacy PBX and Multiple Trixbox
Thu, 02/08/2007 - 1:35pm

Post of copy of the asterisk live log when this call is placed

asterisk -cvvvvvr


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aulax
Posts: 9
 Member Since:
2007-02-08
Re: Routing between Legacy PBX and Multiple Trixbox
Thu, 02/08/2007 - 1:47pm

I made a call from ext. 5205 (Definity) to ext. 3650 (Trixbox - Remote Office):

Feb 8 15:45:12 VERBOSE[5592] logger.c: -- Executing Set("Zap/1-1", "DID=3650") in new stack
Feb 8 15:45:12 VERBOSE[5592] logger.c: -- Executing Goto("Zap/1-1", "s|1") in new stack
Feb 8 15:45:12 VERBOSE[5592] logger.c: -- Goto (from-zaptel,s,1)
Feb 8 15:45:12 VERBOSE[5592] logger.c: -- Executing NoOp("Zap/1-1", "Entering from-zaptel with DID == 3650") in new stack
Feb 8 15:45:12 VERBOSE[5592] logger.c: -- Executing Ringing("Zap/1-1", "") in new stack
Feb 8 15:45:12 DEBUG[5592] chan_zap.c: Requested indication 3 on channel Zap/1-1
Feb 8 15:45:12 DEBUG[5592] pbx.c: Expression result is '0'
Feb 8 15:45:12 DEBUG[5592] pbx.c: Function result is '3650'
Feb 8 15:45:12 VERBOSE[5592] logger.c: -- Executing Set("Zap/1-1", "DID=3650") in new stack
Feb 8 15:45:12 VERBOSE[5592] logger.c: -- Executing NoOp("Zap/1-1", "DID is now 3650") in new stack
Feb 8 15:45:12 DEBUG[5592] pbx.c: Expression result is '1'
Feb 8 15:45:12 VERBOSE[5592] logger.c: -- Executing GotoIf("Zap/1-1", "1?zapok:notzap") in new stack
Feb 8 15:45:12 VERBOSE[5592] logger.c: -- Goto (from-zaptel,s,8)
Feb 8 15:45:12 VERBOSE[5592] logger.c: -- Executing NoOp("Zap/1-1", "Is a Zaptel Channel") in new stack
Feb 8 15:45:12 VERBOSE[5592] logger.c: -- Executing Set("Zap/1-1", "CHAN=1-1") in new stack
Feb 8 15:45:12 DEBUG[5592] pbx.c: Function result is '1'
Feb 8 15:45:12 VERBOSE[5592] logger.c: -- Executing Set("Zap/1-1", "CHAN=1") in new stack
Feb 8 15:45:12 VERBOSE[5592] logger.c: -- Executing Macro("Zap/1-1", "from-zaptel-1|3650|1") in new stack
Feb 8 15:45:12 WARNING[5592] app_macro.c: No such context 'macro-from-zaptel-1' for macro 'from-zaptel-1'
Feb 8 15:45:12 VERBOSE[5592] logger.c: -- Executing NoOp("Zap/1-1", "Returned from Macro from-zaptel-1") in new stack
Feb 8 15:45:12 VERBOSE[5592] logger.c: -- Executing Goto("Zap/1-1", "from-pstn|3650|1") in new stack
Feb 8 15:45:12 VERBOSE[5592] logger.c: -- Goto (from-pstn,3650,1)
Feb 8 15:45:12 WARNING[5592] pbx.c: Channel 'Zap/1-1' sent into invalid extension '3650' in context 'from-pstn', but no invalid handler
Feb 8 15:45:12 DEBUG[5183] channel.c: Avoiding initial deadlock for 'Zap/1-1'
Feb 8 15:45:12 DEBUG[5183] channel.c: Avoiding initial deadlock for 'Zap/1-1'
Feb 8 15:45:12 DEBUG[5183] channel.c: Avoiding initial deadlock for 'Zap/1-1'
Feb 8 15:45:12 DEBUG[5183] channel.c: Avoiding initial deadlock for 'Zap/1-1'
Feb 8 15:45:12 DEBUG[5183] channel.c: Avoiding initial deadlock for 'Zap/1-1'
Feb 8 15:45:12 DEBUG[5183] channel.c: Avoiding initial deadlock for 'Zap/1-1'
Feb 8 15:45:12 DEBUG[5592] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
Feb 8 15:45:12 DEBUG[5592] cdr_addon_mysql.c: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid) VALUES ('2007-02-08 15:45:12','Innes, Fred','','3650','from-pstn', 'Zap/1-1','','Goto','from-pstn|3650|1',0,0,'NO ANSWER',3,'','1170967508.18')
Feb 8 15:45:12 DEBUG[5592] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/1-1
Feb 8 15:45:12 DEBUG[5592] chan_zap.c: Hangup: channel: 1 index = 0, normal = 15, callwait = -1, thirdcall = -1
Feb 8 15:45:12 DEBUG[5592] chan_zap.c: Not yet hungup... Calling hangup once with icause, and clearing call
Feb 8 15:45:12 DEBUG[5592] chan_zap.c: disabled echo cancellation on channel 1
Feb 8 15:45:12 DEBUG[5592] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1
Feb 8 15:45:12 DEBUG[5592] chan_zap.c: Updated conferencing on 1, with 0 conference users
Feb 8 15:45:12 DEBUG[5592] chan_zap.c: Set option AUDIO MODE, value: OFF(0) on Zap/1-1
Feb 8 15:45:12 DEBUG[5592] chan_zap.c: disabled echo cancellation on channel 1
Feb 8 15:45:12 VERBOSE[5592] logger.c: -- Hungup 'Zap/1-1'


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dickson
Posts: 1831
 Member Since:
2006-06-02
Re: Routing between Legacy PBX and Multiple Trixbox
Thu, 02/08/2007 - 2:09pm

Here is something to try:

in your zapata.conf file, you probably have context=from-zaptel

try from-trunk or from-pstn and see if you have any improvements


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aulax
Posts: 9
 Member Since:
2007-02-08
Re: Routing between Legacy PBX and Multiple Trixbox
Thu, 02/08/2007 - 2:34pm

Tried both changes, got fast busy when attempting to dial from an Avaya extension to a Trixbox extension through the Trixbox bridge.

Will the Trixbox bridge automatically route a call to an extension on another Trixbox? I have the Bridge connected to the other Trixbox via SIP trunks. I am calling from an Avaya extension via a T1 tie line into the Bridge Trixbox. Hope that makes sense.


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dickson
Posts: 1831
 Member Since:
2006-06-02
Re: Routing between Legacy PBX and Multiple Trixbox
Thu, 02/08/2007 - 3:26pm

disappointing, had a similar issue and was able to correct it with that change. I assume the call is still arriving?

I'm not 100% sure that I understand what you mean by a bridge in this case.


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kalebksp
Posts: 116
 Member Since:
2006-10-08
Re: Routing between Legacy PBX and Multiple Trixbox
Thu, 02/08/2007 - 4:01pm

I'm no expert so if this totally wrong someone please correct me, but try changing the context to "from-internal" and see if that makes a difference.


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aulax
Posts: 9
 Member Since:
2007-02-08
Re: Routing between Legacy PBX and Multiple Trixbox
Thu, 02/08/2007 - 4:43pm

OK, got a little further. If I had the following to the FROM-ZAPTEL portion of the extensions.conf file:

exten => _36xx,1,Dial(${TRUNK}/${EXTEN})

I now get "The number you have dialed is not in service..." from the Trixbox when I dial from an Avaya extension. How do I get the Trixbox to dial 36xx to a SIP trunk in order to complete the call?


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dickson
Posts: 1831
 Member Since:
2006-06-02
Re: Routing between Legacy PBX and Multiple Trixbox
Thu, 02/08/2007 - 6:32pm

did you try kalebksp's suggestion, put in from-internal and see what it does?


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aulax
Posts: 9
 Member Since:
2007-02-08
Re: Routing between Legacy PBX and Multiple Trixbox
Fri, 02/09/2007 - 9:10am

Finally got it to work by changing the context to FROM-INTERNAL, had to play around to ID the correct group to assign the context. I thank you for the support and quick repsonce to my questions.


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